Definitely a control surface! You are not going to get any decent mixing console in that price range (multiple your budget with 10 and you may have some options). And within your budget you already have chosen the right one: the B-brand thing.

1. Why are you using ASIO4ALL and not UR22's native driver?2. There is no commercial 32-bit a/D or D/A converters. If driver let's you choose "32-bit" option it only means the bit reduction is done inside the driver/digital hardware instead of your DAW (which may or may not be better than reduction inside the DAW depending on driver/hardware/usage profile).

Rename your project folder on your desktop (or move it somewhere else) and copy project folder from the laptop to your desktop. After checking the project from laptop is running smoothly you can delete the old (renamed, moved) folder.

My signature is just kind of a joke. After finding out how some people put all insignificant pieces of hardware/software there, I decided to make my signature as complete as possible ... as you may find out if you read the last line

I often use Melodyne for vocal harmonies ... but only for testing purpose: check out how some kind of harmony would sound and after making decisions just ditch Melodyne harmonies and sing them myself.

But when working with talented singers (that's anyone else that myself) I try to avoid using Melodyne alltogether. If they are there behind Microphones I just try to hum them what I would like them to sing ... a lot faster than dragging those blobs up and down on the computer screen.

I have made the (for me) difficult decision to make this my general use home computer as well I can see that. My studio PCs have always also been a general-purpose PCs ... and it hasn't been a problem to me ... just requires extra care when installing any software. But I always use wired ethernet instead of wireless, when possible (have some 100m/300ft of ethernet cables in my house). 2 reasons for this:1. Every additional hardware/driver is a possible source of problems. So why not use motherboard's ethernet connection and a cable to connect to the router.2. In the past I have had horrible DPC latency problems with numerous wireless devices. So I have decided: no single wireless device connected to my DAW, not even wireless keyboard or mouse, (unless it's DAW-specific device, like my Frotier Tranzport). Maybe this isn't a real problem in these days anymore, but I get paranoid whenever I hear the word "wireless".

I will ask them to look at dropping this for on-board graphics only. It might not be any savings to me to do so It should save you $60

I will inquire about a fanless PSU ... thank you. Should cost you extra $60 (compared to Corsair 500W)

I am not sure what you mean here: That I should look into the case PLUS some acoustic treatment? Or ... This is standard option for ADK computers ... I don't know who is your dealer, but your proposed system is obviously a standard ADK setup: http://www.adkproaudio.com/systems/viewsystem.cfm?recordid=134(under "Quieting/ Xtra cooling" there).It costs extra $100 (compared to your proposed "ADK Quiet Case fans" option)... so it's a bit expensive option, but in my opinion you can't spend too much for making your PC silent. I just built one couple of weeks ago and I'm really happy with the noise level (after spending quite a bit for that), but can't stop thinking: "I should have chosen that fanless PSU and more quiet case fans" (yes, I have acoustically treated case).

What is it that an i7 is better at ... higher track count, or is it more VSTi's? Both and maybe also possible to use lower latency setting. As seen from ADK page (see link above) you can have i7 model 4771 for extra $112. To compare performance difference between 4570 & 4771 see:http://www.cpubenchmark.net/high_end_cpus.htmlIn my opinion the difference is significant and well worth the extra money.

I have a feeling they won't be able to do anything but OEM, but I can always ask! Looks like they have an option of W7 Home Premium (non-OEM) instead of W7 Pro OEM ... and it's even $35 cheaper ... but you have to consider what you loose when having "Home" instead of "Pro". For me it looks like the only significant difference on DAW usage is the "Home" only supporths 16G of RAM memory.

So this is marketing brand of Neil Young's "high-definiton" audio? Poor old Neil. Either he've been misguided or is trying to do the largest scam in the history of music industry. I think I already linked this one: https://people.xiph.org/~xiphmont/demo/neil-young.html the last time this was discussed. No, I'm not against uncompressed/losslessly compressed audio, but (when it comes to delivery format) anything above 44.1k/16 is just snake oil.

In which stage it gets distorted?1. Mic? - If here, your only option is to get him further away from the microphone or change the mic (SM7 should work)2. Preamp? - If here, just turn gain down. If gain is already all the way down, look solution from #1.3. A/D conversion? - Turn down output level of your preamp.4. Mixing? - This is the only stage any plugin can make difference. Any limiter or compressor with high ratio should do the job.

Anyway, only real solution is to give this poor guy a couple of singing lessons.

Not sure if there is Control Room feature in Cubase 'Studio' version. If quite sure it's 'Cubase Only' feature, which means it's in full version Cubase only. Anyway, if it's there, you should find 'Enable Control Room' button in VST Connections window.

A Yamaha digital piano for $700......very nice, 88 keys weighted, 14 yamaha sounds, USB output (can i plug that into the computer or do i use analogue into the interface and then to computer?) While you don't give exact model of your Yamaha keyboard, it's hard to tell ... but probably USB only carries MIDI data and if you want to record Yamaha's audio, you need to connect it's analog outputs to your soundcard/audio interface.

I'm a little without knowledge about how the SSD works.....

As Strophoid already said, it works just like a hyper-fast hard drive.

How much can a 32GB SSD serve up?

To be honest ... it's quite small in today's standards. It was OK 6 years ago (when I bought my first SSD) to hold Windows XP and your basic software, but with Win7 (or 8) and modern software packages it requires some planning (or even serious IT skills) to fit everything into 32G.

Now Monitors baffle me !!!!....i've got $460 for a pair......looking at the Yamaha 5 inchers....8 inchers

If you're looking at primary monitors at this price range I would recommend Adams or KRKs instead of Yamaha.

Why do you dislike USB mic's please? Because they usually have no "real" ASIO drivers and you are forced to use ASIO4ALL. This is a potential source of many kinds of problems.

The Blue Yeti Pro has both USB and XLR.

I see. Forgive me not making enough research. Just read "USB" in specs and immediately freaked out ;)

So you still can plug it into your CI12, which is good, but still you pay extra for USB feature. You could get better mic with same $$$ without USB. I can understand mic like Yeti Pro, if you want to use it also with computer without audio interface (occassional on-site recordings with laptop etc). In that case it even sounds a good idea.

Because my Soundcard (Steinberg CI12+) only has 2 x In/Out :-

This is not excuse IMO. If you need more than 2 inputs at once, USB mic is a bad solution, since once again you'll fall into ASIO4ALL land, which IMO isn't solution for pro DAW.

So you are saying ASIO4ALL is to be avoided at all costs. No. I'm just saying the "real" ASIO driver is better option and you should avoid ASIO4ALL if it's not too difficult/costly.

I thought it would be quite useful to have a mic which had both USB and XLR function? As I said, I too find it useful, if you are going to do recordings on locations where you don't have your audio interface available.

Have to leave now and go to help my niece. She is experiencing audio sync problems when trying to do narrations on her YouTube videos. You may guess which kind of recording hardware system she has. Can you?

Whenever you are "going into red" in Cubase you have already clipped in your Analog/Digital converter. Cubase (or any DAW) can't do anything about it. Only way to resolve the problem is to lower input volume on your instrument, preamp or audio interface.

Where is "I don't care" option. That would be my answer. I prefer to bounce to new track so I have total control on what's happening and not let the software decide. But at the same time if "bounce-in-place" were there, it would not disturb me. But because I have no such option, I have to vote "no". Sorry.

In most cases this problem is because you have used EQ on Windows Media Player to make things sound the way you like. Now, when you do mixdown in Cubase you do the same in Cubase (with Cubase EQ) and AGAIN in Media Player (Media Player EQ): that's doing the same operation twice.

If this is your case, the only solution is to disable Windows Media Player's EQ. If commercially produced music sounds crap after this, you should invest in better headphones/monitors and/or learn to listen to the music way it is (not the way it appears after modifying it with an EQ to suit your taste).

EDIT: "Media Player's EQ" may be substituted by "Sound card driver's EQ" in this reply.

The new audio engine in Cubase 8 are new, and are complete rebuild from scratch in 192kHz, 32bit floating. Cubase audio engine is 192k only if you set your project to 192k. And all correctly implemented digital audio engines sound the same. Period. Fact. If they sound different, there's something seriously wrong with one. When it comes to stock plugins etc, that's a different story, though.

when 4 professionals (the chief mastering engineer mastered more than 75.000 tracks) independently come to a conclusion that there is a evident and pretty big decline in sound on cubase 8 pro. Yes and there's world-class producers (with gazillions #1 hits under their belt) who thinks SATA drive sounds better than USB drive and similar stuff. Go figure.

As seen on the Block Diagram of MG166CX, it's USB only provides 2 channels of audio and these 2 channels are hardwired to STEREO (L/R) buss (pre-fader). If you want to record your guitar and vocals on the different tracks in Cubase, you have to pan them hard left/right in your MG.

This is common (and unfortunate) feature of most cheapish mixing desks with USB interface. You're not the only one who bought this kind of mixer only to find out it can't send individual channels to DAW.

1st - any difference between this "academic" version and the real thing? No. 2nd - if he get's this version does it have limitations on what he can do with what he produces with this version? e.g. can't copyright or sell his music? He can't sell music produced with educational version. Of course he can copyright, because copyright doesn't apply to final product (produced in Cubase) but the music itself (the "idea") and Steinberg (or any other party) can not have any control on that. 3rd - if I buy it for him from here (US) can it be delivered to him in Canada? Shouldn't be a problem.

In any case, please see the educational product FAQ:http://www.steinberg.net/en/education/faq.html

There is no way to change it? am I stuck with it? Please consult your mixer's manual, especially the Block Diagram. It should tell how signals can be routed. Unfortunately I'm not going to do this for you because of one unfortunate thing: F*cking Alesis requires me to enter personal information to their website in order to download the manual. I'm not going to do that.

So, what are the actual pros and cons of recording at higher sample rates Pros:1. In case you have non-oversampling A/D and D/A coverters with poor anti-aliasing/anti-imaging filters the sound quality may improve slightly (flatter freq response near 20kHz and/or less aliasing distortion).2. Some DSP algorithms may introduce aliasing distortion if not implemented correctly. So if you have ill-behaving plugins you can move artifacts they are creating to inaudible frequency range by using higher sample rates (and then removing them completely by downsampling with good quality sample rate converter).

#1 hould be non-issue nowadays (for last 10 - 20 years now depending on if you use pro or consumer grade converters). However #2 may still be valid point today.

I'm just wondering how the HD 192 fares in quality and whether it would be a good idea to Record at higher sample rates? Sorry, I can't tell for sure. I have no experience on HD 192. But it's modern pro/semi-pro unit and should be OK and not having problems of early cheap systems. According this article: http://www.soundonsound.com/sos/oct03/articles/motu.htm HD 192's converters are 128x oversampling. This should eliminate all aliasing/imaging problems in A/D/A conversion. It's probable you won't gain anything by using higher sample rates (when it comes to A/D/A conversion).

Recording in 32-bit floating point is only benefitial if you are going to do a lot, actually a LOT, of offline processing. Or LOT of bouncing.

Fair enough. But then we should suggest Steinberg to implement standard Windows-style menu on Cubase for Mac. Not the other way around as they did. There are many more Windows users than Mac users after all.

This seems to describe how to write the functionality into the program (like Cubase must have done). I certainly don't have the know-how to "insert" this function into the Guitar Rig program. That was my point exactly (should have included a smiley on my post). This behaviour must be programmed inside the program. As a user there nothing you can do, unless you find a third-party program which does something like:1. runs on the background and anytime it sees GuitarRig running turns off power savingor2. turns off power saving functions, runs GuitarRig and turns them back on after GuitarRig has ended

Now after writing this answer, I realised it shouldn't be too difficult to do it yourself:

1. Set you PC into normal operation power scheme (on Control Panel)2. Run CMD and in command prompt typepowercfg /getactivescheme3. write down the resulting GUID string (let's call it GUID2)4. Set you PC into GuitarRig operation power scheme (on Control Panel)5. Run CMD and in command prompt typepowercfg /getactivescheme6. write down the resulting GUID string (let's call it GUID1)

When looking at the Infinite Wave graphs please spend a moment trying to understand them. Distortion figures of all the converters shown there are completely inaudible, except for some of them which may create just audible distortion in very low signal levels. (Cubase is not one of them. It handles low signal levels well.)

When it comes to aliasing, even Cubase's SRC's filter goes down to -60dB at 24kHz which is about the point it starts to matter (aliasing to under 20kHz). You'd need horrible amount of ultrasonic content in your audio to make it's aliasing effect audible.

I use Voxengo r8brain myself, just to feel better, because I have yet to notice any sound difference. Just for the same reason I use SoX on final masters. For "pre-release" stuff I use Cubase.

Thank you, I have done that. I didn't think my post gave the opposite impression, if it did, please understand that was an error on my part. It wasn't response to you, but a general notification. If it looked like an attack against you, I'm very sorry.

Yes, but Cubase's SRC transition filter is shown to only go to -6 dBFS to -12 dBFS at 22-23 kHz. I could easily see that possibly causing audible alias bands. But it doesn't. When converting into to 44.1kHz sample rate, audio in 22-23kHz range aliases into 21-22kHz range. I cannot consider it audible (unless we start talking about intermodulation distortion generated by the analog audio chain). There is a very good reason, why original CD audio standard is 44.1kHz instead of 40kHz: insurance against non-perfect anti-alias filter.

To me, the question remains: How do we know if these graphically obvious differences from ideal, and between different DAWs, are audible? We know if we stop for a moment to think about basic physiology of human hearing.EDIT: just to clarify:1. It's absolutely impossible to hear distortion below -60dB of current signal level 2. We cannot hear anything above 20kHz (for me as an old relic this is more like 15kHz)

Just for the same reason I use SoX on final masters. For "pre-release" stuff I use Cubase. Why not use Cubase on final masters, as you make a strong case that any artifacts of Cubase's SRC are inaudible? For the exactly same reason as peakae does use r8brain: just to feel better. Not having to worry about some exreamly unlikely situation that I have not taken into consideration. Just like:a. I record (and mix and master) at 88.2kHz even though 44.1kHz is fine (just in case there are bad behaving DSP algorithms in my signal chain)b. I do dither when reducing bit depth even though none of the music I produce requires it (because it doesn't cost anything)c. I record 24bit audio even though at least in 99% of my recordings 16 bits will capture all the details (so I don't have to worry about those 1% of cases)

The math is much easier. In matter of fact it isn't. You can't just play "connect-the-dots" game with digital audio. Every time you make sample rate conversion, you have to 1. reconstruct the original audio waveform represented by samples2. pick new samples from reconstructed waveOf course in real life:1. you only have to create a mathematical representation of the original wave (sinc IR response of samples)2. you pick new samples from this function

But this is not that much easier when using multiple sample rates than when using arbitrary sample rates (upsampling is slightly faster but downsampling is just as hard).

IMO we should be more worried about plug-ins doing sampling rate conversions than DAW's off-line conversion algorihtms. Plug-ins must do this in real-time, which is much harder task after all.

with no filtering (a 0dBFS response) at 21 kHz, and only -6dBFS filtering at 22 kHz, how audible would the resulting 1kHz summation tone be for that high-frequency source recorded at high levels? OK! You went into intermodulation distortion path :roll: ... because "Summation tone" of 22 & 21kHz (1kHz) is intermodulation distortion. But the thing is: intermodulation distortion doesn't happen in digital domain, so you can ignore it. It doesn't happen. Only worry we should ever have on Cubase's very shallow anti-alias filter (when comparing to others) is:

* Cubase's filter isn't very effective between 22-24kHz* If we have extreamly powerful audio content in this range (no natural instruments have, very few microphones captures it anyway and any well-programmed VSTi also doesn't produce it) this content aliases into 20kHz-22kHz range* In analog audio hardware (D/A converters, amplifiers, speakers) this aliased audio creates intermodulation distortion in audible range (20Hz-20kHz)BUT:* How much intermodulation distortion is already present produced by legitime audio content (20Hz-20kHz)? Probably something like 1000 times more. CONCLUSION:This is not an issue, except if your music is some kind of strange extreamly high-to-ultra-frequency rich stuff without natural low-to-mid frequency content.

transport show midi activity, when I hit stop the midi line clears Are you saying whenever you're recording, Cubase is drawing a new MIDI part into the track, but when hitting stop, it disappears. This happens if there is nothing recorded to the part. But because you have activity in transport, there should be something coming in. I can only think about two possible reasons:1. You have not selected the right MIDI input for your track2. There's MIDI filtering going on somewhere (eg File | Preferences ... MIDI Filter)